In {{ic|alsamixer}}, the {{ic|MM}} label below a channel indicates that the channel is muted, and {{ic|00}} indicates that it is open.

In {{ic|alsamixer}}, the {{ic|MM}} label below a channel indicates that the channel is muted, and {{ic|00}} indicates that it is open.

−

Scroll to the {{ic|Master}} and {{ic|PCM}} channels with the {{keypress|←}} and {{keypress|→}} keys and unmute them by pressing the {{keypress|m}} key. Use the {{keypress|↑}} key to increase the volume and obtain a value of {{ic|0}} dB gain. The gain can be found in the upper left next to the {{ic|Item:}} field. Higher values of gain will produce distorted sound.

+

Scroll to the {{ic|Master}} and {{ic|PCM}} channels with the {{ic|←}} and {{ic|→}} keys and unmute them by pressing the {{ic|m}} key. Use the {{ic|↑}} key to increase the volume and obtain a value of {{ic|0}} dB gain. The gain can be found in the upper left next to the {{ic|Item:}} field. Higher values of gain will produce distorted sound.

To get full 5.1 or 7.1 surround sound you likely need to unmute other channels such as Front, Surround, Center, LFE (subwoofer) and Side (these are the names of the channels with Intel HD Audio, they may vary with different hardware). Please take note that this will not automatically upmix stereo sources (like most music). In order to accomplish that, see [[#Upmixing/Downmixing]].

To get full 5.1 or 7.1 surround sound you likely need to unmute other channels such as Front, Surround, Center, LFE (subwoofer) and Side (these are the names of the channels with Intel HD Audio, they may vary with different hardware). Please take note that this will not automatically upmix stereo sources (like most music). In order to accomplish that, see [[#Upmixing/Downmixing]].

−

To enable your microphone, switch to the Capture tab with {{keypress|F4}} and enable a channel with {{Keypress|Space}}.

+

To enable your microphone, switch to the Capture tab with {{ic|F4}} and enable a channel with {{ic|Space}}.

−

Leave alsamixer by pressing {{Keypress|Esc}}.

+

Leave alsamixer by pressing {{ic|Esc}}.

{{Note|

{{Note|

Line 105:

Line 105:

</nowiki>}}

</nowiki>}}

−

Use {{ic|<nowiki>$ lsmod | grep snd</nowiki>}} to get a devices list. This configuration assumes you have one mia sound card using {{ic|snd_mia}} and one (e.g. onboard) card using {{ic|snd_hda_intel}}.

+

Use {{ic|<nowiki>$ cat /proc/asound/modules</nowiki>}} to get the loaded sound modules and their order. This list is usually all that is needed for the loading order. Use {{ic|<nowiki>$ lsmod | grep snd</nowiki>}} to get a devices & modules list. This configuration assumes you have one mia sound card using {{ic|snd_mia}} and one (e.g. onboard) card using {{ic|snd_hda_intel}}.

You can also provide an index of {{ic|-2}} to instruct ALSA to never use a card as the primary one. Distributions such as Linux Mint and Ubuntu use the following settings to avoid USB and other "abnormal" drivers from getting index {{ic|0}}:

You can also provide an index of {{ic|-2}} to instruct ALSA to never use a card as the primary one. Distributions such as Linux Mint and Ubuntu use the following settings to avoid USB and other "abnormal" drivers from getting index {{ic|0}}:

Line 168:

Line 168:

pcm.!default {

pcm.!default {

type hw

type hw

−

card 0

+

card 2

}

}

ctl.!default {

ctl.!default {

type hw

type hw

−

card 0

+

card 2

}

}

}}

}}

Line 720:

Line 720:

install snd_hda_intel /bin/false

install snd_hda_intel /bin/false

−

If both devices use the same module, it might be possible to disable one of them in the BIOS.

+

If both devices use the same module then we can use the *enable* parameter from snd-hda-intel module, it's an array of booleans that can enable/disable the desired sound card.

If you use a laptop, pm-utils will change {{ic|power_save}} back to 1 when you go onto battery power even if you disable power saving in {{ic|/etc/modprobe.d}}. Disable this for pm-utils by disabling the script that makes the change (see [https://wiki.archlinux.org/index.php/Pm-utils#Disabling_a_hook Disabling a hook] for more information):

+

If you use a laptop, pm-utils will change {{ic|power_save}} back to 1 when you go onto battery power even if you disable power saving in {{ic|/etc/modprobe.d}}. Disable this for pm-utils by disabling the script that makes the change (see [[Pm-utils#Disabling_a_hook|Disabling a hook]] for more information):

# touch /etc/pm/power.d/intel-audio-powersave

# touch /etc/pm/power.d/intel-audio-powersave

Line 966:

Line 971:

After the changes are loaded successfully, you will see a {{ic|Pre-Amp}} section in alsamixer. You can adjust the levels there.

After the changes are loaded successfully, you will see a {{ic|Pre-Amp}} section in alsamixer. You can adjust the levels there.

{{Note|Setting a high value for {{ic|Pre-Amp}} can cause sound distortion, so adjust it according to the level that suits you.}}

{{Note|Setting a high value for {{ic|Pre-Amp}} can cause sound distortion, so adjust it according to the level that suits you.}}

+

+

{{Note|Some audio codecs may need to have settings adjusted in the HDA Analyzer ([[#HDA analyzer|see above]]) in order to achieve proper volume without distortion. Checking the HP option under widget control in the Playback Switch (Node[0x14] PIN in the ALC892 codec, for instance) can sometimes improve audio quality and volume significantly.}}

=== Popping sound after resuming from suspension ===

=== Popping sound after resuming from suspension ===

Revision as of 00:45, 27 September 2013

zh-CN:Advanced Linux Sound Architecture
The Advanced Linux Sound Architecture (ALSA) is a Linux kernel component which replaced the original Open Sound System (OSSv3) for providing device drivers for sound cards. Besides the sound device drivers, ALSA also bundles a user space library for application developers who want to use driver features with a higher level API than direct interaction with the kernel drivers.

Users with a local login (at a virtual terminal or a display manager) have permission to play audio and change mixer levels. To allow this for a remote login, the user has to be added to the audio group. Membership in the audio group also allows direct access to devices, which can lead to applications grabbing exclusive output (breaking software mixing) and breaks fast-user-switching, and multiseat. Therefore, adding a user to the audio group is not recommended, unless you specifically need to[1].

If you want OSS applications to work with dmix (software mixing), also install the alsa-oss package. Load the snd_seq_oss, snd_pcm_oss and snd_mixer_osskernel modules to enable the OSS emulation modules.

Unmuting the channels

The current version of ALSA installs with all channels muted by default. You will need to unmute the channels manually.

It is easiest to use alsamixer ncurses UI to accomplish this:

$ alsamixer

Alternatively, use amixer from the command-line:

$ amixer sset Master unmute

In alsamixer, the MM label below a channel indicates that the channel is muted, and 00 indicates that it is open.

Scroll to the Master and PCM channels with the ← and → keys and unmute them by pressing the m key. Use the ↑ key to increase the volume and obtain a value of 0 dB gain. The gain can be found in the upper left next to the Item: field. Higher values of gain will produce distorted sound.

To get full 5.1 or 7.1 surround sound you likely need to unmute other channels such as Front, Surround, Center, LFE (subwoofer) and Side (these are the names of the channels with Intel HD Audio, they may vary with different hardware). Please take note that this will not automatically upmix stereo sources (like most music). In order to accomplish that, see #Upmixing/Downmixing.

To enable your microphone, switch to the Capture tab with F4 and enable a channel with Space.

Leave alsamixer by pressing Esc.

Note:

Some cards need to have digital output muted/turned off in order to hear analog sound. For the Soundblaster Audigy LS mute the IEC958 channel.

Some machines, (like the Thinkpad T61), have a Speaker channel which must be unmuted and adjusted as well.

Some machines, (like the Dell E6400) may also require the Front and Headphone channels to be unmuted and adjusted.

You might need to activate the ALSA output in your audio software as well.

Set the default sound card

If your sound card order changes on boot, you can specify their order in any file ending with .conf in /etc/modprobe.d (/etc/modprobe.d/alsa-base.conf is suggested).
For example, if you want your mia sound card to be #0:

Use $ cat /proc/asound/modules to get the loaded sound modules and their order. This list is usually all that is needed for the loading order. Use $ lsmod | grep snd to get a devices & modules list. This configuration assumes you have one mia sound card using snd_mia and one (e.g. onboard) card using snd_hda_intel.

You can also provide an index of -2 to instruct ALSA to never use a card as the primary one. Distributions such as Linux Mint and Ubuntu use the following settings to avoid USB and other "abnormal" drivers from getting index 0:

Select the default PCM via environment variable

You need to replace the default line with the name of your card (in the example is Audigy2). You can get the names with aplay -l or you can also use PCMs like surround51. But if you need to use the microphone it is a good idea to select full-duplex PCM as default.

Now you can start programs selecting the sound card just changing the environment variable ALSAPCM. It works fine for all program that do not allow to select the card, for the others ensure you keep the default card.
For example, assuming you wrote a downmix PCM called mix51to20 you can use it with mplayer using the commandline ALSAPCM=mix51to20 mplayer example_6_channel.wav

Alternative method

First you will have to find out the card and device id that you want to set as the default by running aplay -l:

For example, the last entry in this list has the card ID 2 and the device ID 0. To set this card as the default, you can either use the system-wide file /etc/asound.conf or the user-specific file ~/.asoundrc. You may have to create the file if it does not exist. Then insert the following options with the corresponding card.

pcm.!default {
type hw
card 2
}
ctl.!default {
type hw
card 2
}

The 'pcm' options affect which card and device will be used for audio playback while the 'ctl' option affects which card is used by control utilities like alsamixer .

The changes should take effect as soon as you (re-)start an application (mplayer etc.). You can also test with a command like aplay.

aplay -D default <your_favourite_sound.wav>

If you receive an error regarding your asound configuration, check the upstream documentation for possible changes to the config file format.

Making sure the sound modules are loaded

You can assume that udev will autodetect your sound properly. You can check this with the command

System-wide equalizer

Using AlsaEqual (provides UI)

Note: If you have a x86_64-system and are using a 32bit-flashplugin the sound in flash will not work. Either you have to disable alsaequal or build alsaequal for 32bit.

After installing the package, insert the following into your ALSA configuration file (~/.asoundrc or /etc/asound.conf):

ctl.equal {
type equal;
}
pcm.plugequal {
type equal;
# Modify the line below if you do not
# want to use sound card 0.
#slave.pcm "plughw:0,0";
#by default we want to play from more sources at time:
slave.pcm "plug:dmix";
}
#pcm.equal {
# If you do not want the equalizer to be your
# default soundcard comment the following
# line and uncomment the above line. (You can
# choose it as the output device by addressing
# it with specific apps,eg mpg123 -a equal 06.Back_In_Black.mp3)
pcm.!default {
type plug;
slave.pcm plugequal;
}

And you are ready to change your equalizer using command

$ alsamixer -D equal

Note that configuration file is different for each user (until not specified else) it is saved in ~/.alsaequal.bin.
so if you want to use AlsaEqual with mpd or another software running under different user, you can configure it using

# su mpd -c 'alsamixer -D equal'

or for example, you can make a symlink to your .alsaequal.bin in his home...

High quality resampling

When software mixing is enabled, ALSA is forced to resample everything to the same frequency (48000 by default when supported). dmix uses a poor resampling algorithm which produces noticeable sound quality loss.

samplerate_best offers the best sound quality, but you need a decent CPU to be able to use it as it requires a lot of CPU cycles for real-time resampling. There are other algorithms available (samplerate, etc.) but they may not provide much of an improvement over the default resampler.

Warning: On some systems, enabling samplerate_best may cause a problem where you get no sound from flashplayer.

Upmixing/downmixing

Upmixing

In order for stereo sources like music to be able to saturate a 5.1 or 7.1 sound system, you need to use upmixing. In darker days this used to be tricky and error prone but nowadays plugins exist to easily take care of this task. We will use the upmix plugin, included in alsa-plugins.

Then add the following to your ALSA configuration file of choice (either /etc/asound.conf or ~/.asoundrc):

pcm.upmix71 {
type upmix
slave.pcm "surround71"
delay 15
channels 8
}

You can easily change this example for 7.1 upmixing to 5.1 or 4.0.

This adds a new pcm that you can use for upmixing. If you want all sound sources to go through this pcm, add it as a default below the previous definition like so:

pcm.!default "plug:upmix71"

The plugin automatically allows multiple sources to play through it without problems so setting is as a default is actually a safe choice.
If this is not working, you have to setup your own dmixer for the upmixing PCM like this:

Note: This might not be enough to make downmixing working, see [2]. So, you might also need to add pcm.!default "plug:surround51" or pcm.!default "plug:surround40". Only one vdownmix plug can be used; if you have 7.1 channels, you will need to use surround71 instead the configuration above. A good example, which includes a configuration that makes both vdownmix and dmix working, can be found here.

Mixing

Software mixing (dmix)

Note: For ALSA 1.0.9rc2 and higher on analog sound outputs you do not need to setup dmix. Dmix is enabled as default for soundcards which do not support hardware mixing.

If that does not work however, it is a matter of simply creating a .asoundrc file in your home folder with the following contents.

pcm.dsp {
type plug
slave.pcm "dmix"
}

This should enable software mixing and allows more than one application to make use of the soundcard.

For a digital sound output such as S/PDIF, the ALSA package still does not enable dmix by default. Thus, the dmix configuration above can be used to enable dmix for S/PDIF devices.

Hardware mixing

Support

If you have an audio chipset that supports mixing in hardware, then no configuration is necessary. Almost every onboard audio chipset does not support hardware mixing, and requires mixing to be done in software (see above). Many sound cards do support hardware mixing, and the ones best supported on Linux are listed below:

Creative SoundBlaster Live! (5.1 model)

Creative SoundBlaster Audigy (some models)

Creative SoundBlaster Audidy 2 (ZS models)

Creative SoundBlaster Audigy 4 (Pro models)

Note:

The low end variants of above cards, (Audigy SE, Audigy 2 NX, SoundBlaster Live! 24bit and SoundBlaster Live! 7.1) do not support hardware mixing as they use other chips.

The onboard VIA8237 chip supports 4-stream hardware mixing, however, it does only 3 for some motherboards (the 4th makes no sound) or is just broken. Even if it works, the quality is not good compared to other solutions.

Fixes

If you are using 64-bit Arch and the Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02), you can get sound working for Enemy Territory with the following:

Troubleshooting

Sound skipping while using dynamic frequency scaling

Some combinations of ALSA drivers and chipsets may cause audio from all sources to skip when used in combination with a dynamic frequency scaling governor such as ondemand or conservative. Currently, the solution is to switch back to the performance governor.

Problems with availability to only one user at a time

You might find that only one user can use the dmixer at a time. This is probably ok for most, but for those who run mpd as a separate user this poses a problem. When mpd is playing a normal user cannot play sounds though the dmixer. While it's quite possible to just run mpd under a user's login account, another solution has been found. Adding the line ipc_key_add_uid 0 to the pcm.dmixer block disables this locking. The following is a snippet of the asound.conf, the rest is the same as above.

Simultaneous playback problems

If you are having problems with simultaneous playback, and if PulseAudio is installed (i.e. by GNOME), its default configuration is set to "hijack" the soundcard. Some users of ALSA may not want to use PulseAudio and are quite content with their current ALSA settings. One fix is to edit /etc/asound.conf and comment out the following lines:

Timidity

If it failed, try # killall -9 timidity. If this solves the issue, then you should disable the timidity daemon to be started at boot.

Specific program problems

For other programs who insist on their own audio setup, eg, XMMS or Mplayer, you would need to set their specific options.

For mplayer, open up ~/.mplayer/config (or /etc/mplayer/mplayer.conf for global setting) and add the following line:

ao=alsa

For XMMS/Beep Media Player, go into their options and make sure the sound driver is set to Alsa, not oss.

To do this in XMMS:

Open XMMS

Options > Preferences.

Choose the ALSA output plugin.

For applications which do not provide a ALSA output, you can use aoss from the alsa-oss package. To use aoss, when you run the program, prefix it with aoss, eg:

aoss realplay

pcm.!default{ ... } doesnt work for me anymore. but this does:

pcm.default pcm.dmixer

Model settings

Although Alsa detects your soundcard through the BIOS at times Alsa may not be able to recognize your model type. The soundcard chip can be found in alsamixer (e.g. ALC662) and the model can be set in /etc/modprobe.d/modprobe.conf or /etc/modprobe.d/sound.conf. For example:

options snd-hda-intel model=MODEL

There are other model settings too. For most cases Alsa defaults will do. If you want to look at more specific settings for your soundcard take a look at the Alsa Soundcard List find your model, then Details, then look at the "Setting up modprobe..." section. Enter these values in /etc/modprobe.d/modprobe.conf. For example, for an Intel AC97 audio:

Conflicting PC speaker

If you are sure nothing is muted, that your drivers are installed correctly, and that your volume is right, but you still do not hear anything, then try adding the following line to /etc/modprobe.d/modprobe.conf:

options snd-NAME-OF-MODULE ac97_quirk=0

The above fix has been observed to work with via82xx

options snd-NAME-OF-MODULE ac97_quirk=1

The above fix has been reported to work with snd_intel8x0

No microphone input

In alsamixer, make sure that all the volume levels are up under recording, and that CAPTURE is toggled active on the microphone (e.g. Mic, Internal Mic) and/or on Capture (in alsamixer, select these items and press space). Try making positive Mic Boost and raising Capture and Digital levels higher; this make make static or distortion, but then you can adjust them back down once you are hearing something when you record

As the pulseaudio wrapper is shown as "default" in alsamixer, you may have to press F6 to select your actual soundcard first. You may also need to enable and increase the volume of Line-in in the Playback section.

To test the microphone, run these commands (see arecord's man page for further information):

$ arecord -d 5 test-mic.wav
$ aplay test-mic.wav

If all fails, you may want to eliminate hardware failure by testing the microphone with a different device.

For at least some computers, muting a microphone (MM) simply means its input does not go immediately to the speakers. It still receives input.

Many Dell laptops need "-dmic" to be appended to the model name in /etc/modprobe.d/modprobe.conf:

options snd-hda-intel model=dell-m6-dmic

Some programs use try to use OSS as the main input software. If you have enabled the snd_pcm_oss, snd_mixer_oss or snd_seq_osskernel modules previously (they are not loaded by default), try unloading them.

Setting the default microphone/capture device

Some applications (Pidgin, Adobe Flash) do not provide an option to change the capture device. It becomes a problem if your microphone is on a separate device (e.g. USB webcam or microphone) than your internal sound card. To change only the default capture device, leaving the default playback device as is, you can modify your ~/.asoundrc file to include the following:

Replace "U0x46d0x81d" with your capture device's card name in ALSA. You can use arecord -L to list all the capture devices detected by ALSA.

Internal microphone not working

First make sure all the volume levels are up under recording in alsamixer. In my case adding the following option to /etc/sound.conf and reloading the snd-* module produced a new volume setting called Capture which was capturing for the internal mic. For eg, for snd-hda-intel add

options snd-hda-intel enable_msi=1

Then reload the module (as below), up the recording volume of Capture and then test.

# rmmod snd-hda-intel && modprobe snd-hda-intel

No sound with onboard Intel sound card

There may be a problem with two conflicting modules loaded, namely snd_intel8x0 and snd_intel8x0m. In this case, blacklist snd_intel8x0m:

/etc/modprobe.d/modprobe.conf

blacklist snd_intel8x0m

Muting the "External Amplifier" in alsamixer or amixer may also help. See the ALSA wiki.

Unmuting the "Mix" setting in the mixer might help, also.

No headphone sound with onboard Intel sound card

With Intel Corporation 82801 I (ICH9 Family) HD Audio Controller on laptop, you may need to add this line to modprobe or sound.conf:

options snd-hda-intel model=model

Where model is any one of the following (in order of possibility to work, but not merit):

dell-vostro

olpc-xo-1_5

laptop

dell-m6

laptop-hpsense

Note: It may be necessary to put this "options" line below (after) any "alias" lines about your card.

You can see all the available models in the kernel documentation. For example here, but check that it is the correct version of that document for your kernel version.

A list of available models is also available here. To know your chip name type the following command (with * being corrected to match your files). Note that some chips could have been renamed and do not directly match the available ones in the file.

$ grep Codec /proc/asound/card*/codec*

Note that there is a high chance none of the input devices (all internal and external mics) will work if you choose to do this, so it is either your headphones or your mic. Please report to ALSA if you are affected by this bug.

Pops when starting and stopping playback

Some modules (e.g. snd_ac97_codec and snd_hda_intel) can power off your sound card when not in use. This can make an audible noise (like a crack/pop/scratch) when turning on/off your sound card. Sometimes even when move the slider volume, or open and close windows (KDE4). If you find this annoying try modinfo snd_MY_MODULE, and look for a module option that adjusts or disables this feature.

You may also have to unmute the 'Line' ALSA channel for this to work. Any value will do (other than '0' or something too high).

Example: on an onboard VIA VT1708S (using the snd_hda_intel module) these cracks occured even though 'power_save' was set to 0. Unmuting the 'Line' channel and setting a value of '1' solved the problem.

If you use a laptop, pm-utils will change power_save back to 1 when you go onto battery power even if you disable power saving in /etc/modprobe.d. Disable this for pm-utils by disabling the script that makes the change (see Disabling a hook for more information):

# touch /etc/pm/power.d/intel-audio-powersave

S/PDIF output does not work

If the optical/coaxial digital output of your motherboard/sound card is not working or stopped working, and have already enabled and unmuted it in alsamixer, try running

# iecset audio on

You can also put this command in an enabled systemd service as it sometimes it may stop working after a reboot.

HDMI output does not work

The procedure described below can be used to test HDMI audio. Before proceeding, make sure you have enabled and unmuted the output with alsamixer.

Note: If you are using an ATI card and linux kernel >=3.0, a necessary kernel module is disabled by default. See ATI#HDMI_Audio.

Connect your PC to the Display via HDMI cable and enable the display with a tool such as xrandr or arandr. For example:

Send sound to the device. Following the example in the previous step, you would send sound to card 1, device 3:

$ aplay -D plughw:1,3 /usr/share/sounds/alsa/Front_Center.wav

If aplay does not output any errors, but still no sound is heared, "reboot" the receiver, monitor or tv set. Since the HDMI interface executes a handshake on connection, it might have noticed before that there was no audio stream embedded, and disabled audio decoding. In particular, if you are using a standalone window manager (don’t know about Gnome or KDE), you may need to have some sound playing while plugging in the HDMI cable.

If the test is successful, create or edit your ~/.asoundrc file to set HDMI as the default audio device.

~/.asoundrc

pcm.!default {
type hw
card 0
device 3
}

Or you above config does not work try:

~/.asoundrc

defaults.pcm.card 0
defaults.pcm.device 3
defaults.ctl.card 0

HDMI multi-channel PCM output does not work (Intel)

As of Linux 3.1 multi-channel PCM output through HDMI with a Intel card (Intel Eaglelake, IbexPeak/Ironlake,SandyBridge/CougarPoint and IvyBridge/PantherPoint) is not yet supported. Support for it has been recently added and expected to be available in Linux 3.2. To make it work in Linux 3.1 you need to apply the following patches:

Skipping sound when playing MP3

If you have sound skipping when playing MP3 files and you have more then 2 speakers attached to your computer (i.e. > 2 speaker system), run alsamixer and disable the channels for the speakers that you DO NOT have (i.e. do not enable the sound for the center speaker if you do not have a center speaker.

Using a USB headset and external USB sound cards

If you are using a USB headset with ALSA you can try using asoundconfAUR (currently only available from the AUR) to set the headset as the primary sound output. Before running make sure you have usb audio module enabled (modprobe snd-usb-audio).

Hot-plugging a USB sound card

In order to automatically make a USB Sound Card the primary output device, when the card is plugged in, you can use the following udev rules (e.g. add the following two lines to /etc/udev/rules.d/00-local.rules and reboot).

Error 'Unknown hardware' appears after a kernel update

The following messages may be displayed during the start-up ALSA after the kernel update:

Unknown hardware "foo" "bar" ...
Hardware is initialized using a guess method
/usr/bin/alsactl: set_control:nnnn:failed to obtain info for control #mm (No such file or directory)

or:

Found hardware: "HDA-Intel" "VIA VT1705" "HDA:11064397,18490397,00100000" "0x1849" "0x0397"
Hardware is initialized using a generic method
/usr/bin/alsactl: set_control:1328: failed to obtain info for control #1 (No such file or directory)
/usr/bin/alsactl: set_control:1328: failed to obtain info for control #2 (No such file or directory)
/usr/bin/alsactl: set_control:1328: failed to obtain info for control #25 (No such file or directory)
/usr/bin/alsactl: set_control:1328: failed to obtain info for control #26 (No such file or directory)

Simply store ALSA mixer settings again:

# alsactl -f /var/lib/alsa/asound.state store

It may be necessary configure ALSA again with alsamixer

HDA analyzer

If the mappings to your audio pins(plugs) do not correspond but ALSA works fine, you could try HDA Analyzer -- a pyGTK2 GUI for HD-audio control can be found at the ALSA wiki.
Try tweaking the Widget Control section of the PIN nodes, to make microphones IN and headphone jacks OUT. Referring to the Config Defaults heading is a good idea.

Note:

The script is done by such way that it is incompatible with python3 (which is now shipped with ArchLinux) but tries to use it.

The workaround is: open "run.py", find all occurences of "python" (2 occurences - one on the first line, and the second on the last line) and replace them all by "python2".

NOTE2: the script requires root acces, but running it via su/sudo is bogus. Run it via kdesu or gksu.

ALSA with SDL

If you get no sound via SDL and ALSA cannot be chosen from the application. Try setting the environmental variable SDL_AUDIODRIVER to alsa.

# export SDL_AUDIODRIVER=alsa

Low sound workaround

If you are facing low sound even after maxing out your speakers/headphones, you can give the softvol plugin a try. Add the following to /etc/asound.conf.

Note: You will probably have to restart the computer, as restarting the alsa daemon did not load the new configuration for me. Also, if the configuration does not work even after restarting, try changing plug with hw in the above configuration.

After the changes are loaded successfully, you will see a Pre-Amp section in alsamixer. You can adjust the levels there.

Note: Setting a high value for Pre-Amp can cause sound distortion, so adjust it according to the level that suits you.

Note: Some audio codecs may need to have settings adjusted in the HDA Analyzer (see above) in order to achieve proper volume without distortion. Checking the HP option under widget control in the Playback Switch (Node[0x14] PIN in the ALC892 codec, for instance) can sometimes improve audio quality and volume significantly.

Popping sound after resuming from suspension

You might hear a popping sound after resuming the computer from suspension. This can be fixed by editing /etc/pm/sleep.d/90alsa and removing the line that says aplay -d 1 /dev/zero

Mute after reboot

After reboot, sound setting by alsamixer can not restore. Maybe you can restore by command : sudo alsactl restore. Please check the Auto-Mute toggle status in alsamixer : set Enabled to Disabled.